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 To :  <yate@v...>
 From :  "Serge Kruppa" <serge.kruppa@s...>
 Subject :  Yate1 call queuing in H.323 to SIP proxy scenario question
 Date :  Mon, 28 Aug 2006 07:33:35 -0500
Hello,

 

I have to route calls coming from an H.323 Avaya PBX to a SIP-based IVR
system. I have tested Yate1 under Debian in H.323 to SIP proxy mode between
two soft-phones and it worked fine. In order to move to the production
system I need one additional feature: queuing of the H.323 calls with music
on hold (MOH) if the capacity (60 simultaneous calls) of the SIP IVR has
been exceeded. Once an IVR port becomes available, the queued H.323 call
needs to have MOH interrupted and be automatically directed to the SIP IVR.
Statistics (CDRs) must be gathered to measure the call abandon rate, etc.

 

How do I set up call queuing, MOH and statistics in the H.323 to SIP proxy
scenario? Any suggestion would be most appreciated.

 

Kind Regards,

 

Serge

 




Hello,

 

I have to route calls coming from an H.323 Avaya PBX to a SIP-based IVR system. I have tested Yate1 under Debian in H.323 to SIP proxy mode between two soft-phones and it worked fine. In order to move to the production system I need one additional feature: queuing of the H.323 calls with music on hold (MOH) if the capacity (60 simultaneous calls) of the SIP IVR has been exceeded. Once an IVR port becomes available, the queued H.323 call needs to have MOH interrupted and be automatically directed to the SIP IVR. Statistics (CDRs) must be gathered to measure the call abandon rate, etc.

 

How do I set up call queuing, MOH and statistics in the H.323 to SIP proxy scenario? Any suggestion would be most appreciated.

 

Kind Regards,

 

Serge